The present invention relates to an audio signal transmission system having a noise reduction means and a digital signal processor, and in particular, to an audio signal transmission system suitable for reproducing signals with reduced noise and low distortion.
As a method for transmitting signals, a pulse code modulation (PCM) system has been commonly utilized in which the signals to be transmitted are digitalized and undergo pulse code modulation. The PCM system is applicable to various fields because it has such characteristics that a signal can be restored even when the waveform is distorted or any noise is added thereto during the transmission process for transmitting the signal so long as the "0's" and "1's" in the signal can be identified, and that time-base compression and expansion can be effected on the signals without difficulty.
FIG. 1 shows a schematic block diagram depicting the construction of a prior art tranmission system in which pulse code modulation is carried out on the audio signals. Referring to FIG. 1, an audio signal received at an input terminal 50 is passed through a low-pass filter (LPF) 1 which filters out the component of the audio signal whose frequency exceeds half the sampling frequency of an analog-to-digital (A/D) converter 7, and the resultant signal is delivered to the A/D converter 7. This frequency band suppression is performed to prevent a folded noise which may be caused by the component of the audio signal whose frequency is greater than half the sampling signal and is indispensable for a transmitter operating in accordance with the PCM system.
The A/D converter 7 converts the audio signal inputted thereto into a digital signal, which is then applied to a PCM processor 9. In the PCM processor 9, the received digital audio signal is modulated to obtain a PCM signal and time-base compression is effected on the PCM signal, then the error detection/correction codes and sync codes are added thereto and the obtained signal is delivered to a transmission path 60, such as a recording medium.
The PCM audio signal reproduced from the transmission path 60 is fed to a waveform equalizer circuit 12 to remove the interference between signals, then the resultant signal is fed to a data strobe circuit 13. The jitter contained in the reproduced signal due to certain causes, such as fluctuation in the transmission path 60, is removed by the data strobe circuit 13, thereby shaping the waveform to obtain a shaped signal. The PCM audio signal subjected to the waveform shaping is delivered to the PCM processor 9, which then performs thereon the error correction and error concealment, pulse code demodulation, and time-base expansion and transfers the obtained signal to a digital-to-analog (D/A) converter 8. The D/A converter 8 converts the received digital audio signal into an analog audio signal, which is then outputted to a low-pass filter (LPF) 2. Unnecessary high-frequency components, such as those caused by the sampling operation in the A/D converter 7, are removed by the low-pass filter 2 so as to reproduce the original audio signal. The reproduced audio signal is supplied from an output terminal 51 to an audio playback system.
In a transmitter which performs the PCM processing on audio signals as described above, however, a high-quality audio signal cannot be easily attained because noise called quantizing noise inherent to a digital system appears if the number of bits assigned to a sample value of the audio signal in the A/D converter is only around eight. To overcome this difficulty, approximately 14 to 16 bits are required for a sample value, which leads to a drawback that a broad band is necessary for the PCM signal transmission.
A method for solving this problem and for obtaining a high-quality audio signal while assigning only about eight bits to each sample value has been proposed in the Japanese Patent Laid-Open No. 57-129549 laid open on Aug. 11, 1982 in which an analog-signal noise reduction (NR) means is provided in a PCM transmission system as described above. The analog-signal noise reduction means equivalently reduces the quantizing noise taking place in a digital processing system by compressing the dynamic range of the input signal by the use of an amplitude compressor at signal transmission (modulation) and by expanding the dynamic range of the reproduced signal by use of an amplitude expander at signal reproduction (demodulation).
Although the proposal described in the Japanese Patent Laid-Open No. 57-129549 relates to a combination of noise reduction means in the analog and PCM signal transmission systems, it does not touch upon the problem associated with the operating frequency band of the noise reduction means and the transmission band of the PCM signal transmission system.
FIG. 2 illustrates a circuit block diagram of a system comprising the noise reduction means and a transmitter operating in accordance with the PCM system.
Referring to FIG. 2, the audio signal received at the input terminal 50 fed to a compressor 20 which compresses the dynamic range of the input signal, then the obtained signal is passed through the low-pass filter 1 which passes the input signal whose frequency does not exceed half the sampling signal of the PCM transmitter described above and is delivered to the PCM transmission system 30, which corresponds to the system illustrated in FIG. 1. In the playback, on the other hand, a signal delivered from the PCM transmission system 30 is fed to an expander 21 for expanding the dynamic range of the signal through a low-pass filter 2 having the same pass-band as the low-pass filter 1 to restore the original passband, then the resultant signal is fed from the output terminal 51. In FIG. 2, the blocks enclosed within the broken lines represents a noise reduction compander b comprising the range compressor 20 and range expander 21.
However, in a system in which the noise reduction compander and pulse-code modulation-demodulator (modem) are simply combined as depicted in FIG. 2, the distortion rate of the reproduced signal may be considerably deteriorated because the range expander 21 in the noise reduction compander possibly performs incorrect expansion in some cases.
The erroneous operation of the noise reduction means will be described in detail by referring to FIG. 3 to FIG. 5.
FIG. 3a depicts an example of the input/output amplitude characteristic developed by the compressor 20 which operates by detecting the amplitude of the input signal in the full band range, while FIG. 3b illustrates an example of the input/output characteristic of the expander 21 operating in the same manner as the compressor 20. The graph of FIG. 3 represents the relationship between the input and output levels developed by the compression represented by the following expression; ##EQU1## where the reference level of the input signal is assumed to be 0 decibel. The characteristic of FIG. 3b is obtained by executing the following expansion assuming that the reference level of the input signal is 0 decibel: EQU Output level (dB)=2.times.Input level (db) (2)
As an example, consequently, if noise whose magnitude is -40 dB occurs on a playback signal which has undergone a 1/2 compression having the characteristic described above through the signal transmission, the magnitude of the noise can be reduced to -80 dB in accordance with the expansion characteristic of the noise reduction means 6.
However, if the band of the input signal at compression differs from that of the input signal at expansion, an erroneous operation in which signal compression and expansion are not consistent with each other takes place in the compressor/expander circuit because of the different detecting wave bands for the compressor and expander, thereby deteriorating the quality of the playback sound.
In the system comprising the combination of the noise reduction means and PCM transmission system, therefore, if the input signal to the compressor 20 has a band corresponding to the characteristic curve 40 depicted in FIG. 4a and the band of the input signal to the expander 21 is limited as depicted in the characteristic curve 41 of FIG. 4b because it is determined by the transmission band not exceeding half the sampling frequency of the PCM transmission system, the frequency components of the input signal received for the compression and which are indicated by a hatched area 42 in FIG. 4a are missing in the input signal received for the expansion, thereby making it impossible to recreate the original signal correctly. Assume that the input signal to the compressor 20 comprises the two frequency components shown in FIG. 4b in which the high-frequency component has a frequency exceeding half the sampling frequency and the input level thereof is X (dB) with respect to the reference input level 0 dB. The input level of the low-frequency component of the input signal is Y (dB) with respect to the reference input level 0 dB, where Y is less than X. The input signal, in this case, is compressed by the compressor 20 so that the input level X (dB) becomes to be X/2 (dB) before it is transmitted. That is, the gain G (dB) developed through this compression is expressed as: EQU G (dB)=X/2 (dB)-X (dB)=-X/2 (dB) (3)
The compressed signal level Y' (dB) of the low-frequency component is represented as follows: EQU Y' (dB)=Y (dB)+G (dB)=Y (dB)-X/2 (dB) (4)
For the expansion, on the other hand, the high-frequency components exceeding half the sampling frequency are not transmitted through the pulse code modulation/demodulation section, that is, only the signal comprising the low-frequency components is fed to the expander 21. The input signal level of the input signal to the expander 21 is thus expressed to be Y' (dB) by the equation (4) above. The output level Y.sub.PB (dB) of the output signal from the expander 21 is consequently derived from the equation (2) as follows: EQU Y.sub.PB (dB)=2.times.Y' (dB)=2Y (dB)-X (dB) (5)
As can be clear from this result, an erroneous expansion is caused, that is, the input level Y (dB) for the compression canot be recreated and the quality of the reproduced sound is deteriorated.
An example of the erroneous expansion will be described by referring to FIG. 5. The compression characteristic of the 1/2 compression by the compressor 20 is represented by the direct line (1) in FIG. 5, while the expansion characteristic (.times.2) of the expander 21 is indicated by the line (2).
Assume that the input signal level X (dB) (the input level of the signal comprising the above-mentioned two frequency components) of the compressor 20 is -20 dB and that the signal level Y (dB) of the low-frequency component is -25 dB, then the signal levels after the compression are obtained as illustrated in FIG. 5 based on the equations (1) and (5). Namely, the output level of the signal comprising the two frequency components is -10 dB, while the output level Y' (dB) of the low frequency components is -15 dB. For the expansion as described above, on the other hand, the signal received by the expander 21 comprises only the low-frequency components, so the signal level of the input signal to the expandor 21 is -15 dB. Consequently, the expanded output level Y.sub.PB (dB) of the output signal becomes -30 dB as depicted in FIG. 5, thus, the output signal level is not equal to the input level which is -25 dB of the signal received by the compressor 20.
As described above, the quality of the reproduced sound is remarkably deteriorated by the erroneous expansion particularly when the compression of the compressor 20 and the expansion of the expander 21 are restricted by the amplitude detection signal associated with the full-band components of the input signal or at least the high-frequency components thereof.